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Advances in digital speech transmission / / edited by Rainer Martin, Ulrich Heute, Christiane Antweiler
Advances in digital speech transmission / / edited by Rainer Martin, Ulrich Heute, Christiane Antweiler
Pubbl/distr/stampa Chichester, West Sussex, England ; , : John Wiley & Sons, , c2008
Descrizione fisica 1 online resource (573 p.)
Disciplina 621.39/9
Altri autori (Persone) MartinRainer
HeuteUlrich
AntweilerChristiane
Soggetto topico Speech processing systems
Signal processing - Digital techniques
ISBN 1-281-32198-2
9786611321987
0-470-72718-7
0-470-72717-9
Formato Materiale a stampa
Livello bibliografico Monografia
Lingua di pubblicazione eng
Nota di contenuto -- List of Contributors xxi -- Preface xxvii -- 1 Introduction 1 /Rainer Martin, Ulrich Heute, Christiane Antweiler -- I Speech Quality Assessment 7 -- 2 Speech-Transmission Quality: Aspects and Assessment for Wideband vs. Narrowband Signals 9 /Ulrich Heute -- 2.1 Introduction 9 -- 2.2 Speech Signals . 10 -- 2.3 Telephone-Band Speech Signals 11 -- 2.4 Wideband Speech Signals 14 -- 2.5 Speech-Quality Assessment 25 -- 2.6 Wideband Speech-Quality Assessment 30 -- 2.7 Concluding Remarks 43 -- Bibliography 44 -- 3 Parametric Quality Assessment of Narrowband Speech in Mobile Communication Systems 51 /Marc Werner -- 3.1 Introduction 51 -- 3.2 Simulations of GSM and UMTS Speech Transmissions 58 -- 3.3 Speech Quality Measures based on Transmission Parameters 65 -- 3.4 Discussion and Conclusions 73 -- Bibliography 73 -- II Adaptive Algorithms in Acoustic Signal Processing 77 -- 4 Kalman Filtering in Acoustic Echo Control: A Smooth Ride on a Rocky Road 79 /Gerald Enzner -- 4.1 Introduction 79 -- 4.2 A Comprehensive Theory of Acoustic Echo Control 85 -- 4.3 The Kalman Filter for Conditional Mean and Covariance Estimation 90 -- 4.4 AEC Performance of the Frequency-Domain Adaptive Kalman Filter 100 -- 4.5 Discussion and Conclusions 102 -- Bibliography 103 -- 5 Noise Reduction - Statistical Analysis and Control of Musical Noise 107 /Colin Breithaupt, Rainer Martin -- 5.1 Introduction 107 -- 5.2 Speech Enhancement in the DFT Domain 109 -- 5.3 Measurement and Assessment of Unnatural Fluctuations 115 -- 5.4 Avoidance of Processing Artifacts 120 -- 5.5 Control of Spectral Fluctuations in the Cepstral Domain 123 -- 5.6 Discussion and Conclusions 128 -- 5.7 Acknowledgements 129 -- 5.8 Appendix 129 -- Bibliography 131 -- 6 Acoustic Source Localization with Microphone Arrays 135 /Nilesh Madhu, Rainer Martin -- 6.1 Introduction 135 -- 6.2 SignalModel 136 -- 6.3 Localization Approach Taxonomy 141 -- 6.4 Indirect Localization Approaches 141 -- 6.5 Direct Localization Approaches 148 -- 6.6 Evaluation of Localization Algorithms 156.
6.7 Conclusions 166 -- Bibliography 166 / /7 Multi-Channel System Identification with Perfect Sequences / Theory and Applications / 171 /Christiane Antweiler -- 7.1 Introduction 171 -- 7.2 System Identification with Perfect Sequences 174 -- 7.3 Multi-Channel System Identification 185 -- 7.4 Applications 191 -- 7.5 Discussion and Conclusions 195 -- Bibliography 195 -- III Speech Coding for Heterogeneous Networks 199 -- 8 Embedded Speech Coding: From G.711 to G.729.1 201 /Bernd Geiser, Stéphane Ragot, Hervé Taddei -- 8.1 Introduction 201 -- 8.2 Theory and Tools of Embedded Speech Coding 203 -- 8.3 Embedded Speech Coding Methods 212 -- 8.4 Standardized Embedded Speech Coders 219 -- 8.5 Network Aspects of Embedded Speech Coding 232 -- 8.6 Conclusions and Perspectives 237 -- Bibliography 238 -- 9 Backwards Compatible Wideband Telephony 249 /Peter Jax -- 9.1 Introduction 249 -- 9.2 From Narrowband Telephony to Wideband Telephony 250 -- 9.3 Stand-Alone Bandwidth Extension 254 -- 9.4 Embedded Wideband Coding Using Bandwidth Extension Techniques 257 -- 9.5 Combination of Bandwidth Extension with Watermarking 262 -- 9.6 Advanced Transmission of Highband Parameters 267 -- 9.7 Conclusions 274 -- Bibliography 274 -- IV Joint Source-Channel Coding 279 -- 10 Parameter Models and Estimators in Soft Decision Source Decoding 281 /Tim Fingscheidt -- 10.1 Introduction 281 -- 10.2 Overview to Soft Decision Source Decoding 283 -- 10.3 The Markovian Parameter Model 287 -- 10.4 Basic Extrapolative Estimators 290 -- 10.5 Joint Extrapolative Estimation of Two Different Parameters 298 -- 10.6 Extrapolative Estimation with Repeated Parameter Transmission 301 -- 10.7 Interpolative Estimation of a Parameter 304 -- 10.8 Discussion and Conclusions 307 -- Bibliography 307 -- 11 Optimal MMSE Estimation for Vector Sources with Spatially and Temporally Correlated Elements 311 /Stefan Heinen, Marc Adrat -- 11.1 Introduction 311 -- 11.2 Source Model 312 -- 11.3 Transmission Channel 316 -- 11.4 Optimal MMSE Parameter Estimator 316.
11.5 Near-Optimal MMSE Parameter Estimator 320 -- 11.6 Illustrative Comparison 323 -- 11.7 Simulation Results 325 -- 11.8 Conclusions 327 -- Bibliography 327 -- 12 Source Optimized Channel Codes & Source Controlled Channel Decoding 329 /Stefan Heinen, Thomas Hindelang -- 12.1 Introduction 329 -- 12.2 The Transmission System Used as Reference 330 -- 12.3 Source Optimized Channel Coding (SOCC) 332 -- 12.4 Source Controlled Channel Decoding (SCCD) 341 -- 12.5 Comparison of SOCC versus SCCD 357 -- 12.6 Conclusions 362 -- Bibliography 363 -- 13 Iterative Source-Channel Decoding & Turbo DeCodulation 365 /Marc Adrat, Thorsten Clevorn, Laurent Schmalen -- 13.1 Introduction 365 -- 13.2 The Key of the Turbo Principle: Extrinsic Information 366 -- 13.3 Iterative Source-Channel Decoding (ISCD) 379 -- 13.4 Turbo DeCodulation (TDeC) 387 -- 13.5 Conclusions 394 -- Bibliography 395 -- V Speech Processing in Hearing Instruments 399 -- 14 Binaural Signal Processing in Hearing Aids 401 /Volkmar Hamacher, Ulrich Kornagel, Thomas Lotter, Henning Puder -- 14.1 Introduction 401 -- 14.2 Wireless System for Hearing Aids 405 -- 14.3 Binaural Classification Systems 410 -- 14.4 Binaural Beamformer 415 -- 14.5 Blind Source Separation (BSS): An Application for a Binaural Directional Microphone Array in Hearing Aids 422 -- 14.6 Conclusions 427 -- Bibliography 428 -- 15 Auditory-profile-based Physical Evaluation of Multi-microphone Noise Reduction Techniques in Hearing Instruments 431 /Koen Eneman, Arne Leijon, Simon Doclo, Ann Spriet, Marc Moonen, Jan Wouters -- 15.1 Introduction 431 -- 15.2 Multi-microphone Noise Reduction in Hearing Instruments 434 -- 15.3 Auditory-profile-based Physical Evaluation 441 -- 15.4 Test Conditions 449 -- 15.5 Simulation Results 450 -- 15.6 Discussion 452 -- 15.7 Conclusions 455 -- Bibliography 456 -- VI Speech Processing for Human / Machine Interfaces 459 -- 16 Automatic Speech Recognition in Adverse Acoustic Conditions 461 /Hans-GŠunter Hirsch -- 16.1 Introduction 461.
16.2 Structure of Speech Recognition Systems 462 -- 16.3 Acoustic Scenarios during Speech Input 468 -- 16.4 Improving the Recognition Performance in Adverse Conditions 476 -- 16.5 Conclusions 493 -- Bibliography 494 -- 17 Speaker Classification for Next-Generation Voice-Dialog Systems 497 /Felix Burkhardt, Florian Metze, Joachim Stegmann -- 17.1 Introduction 497 -- 17.2 Speaker Classification 498 -- 17.3 Detection of Age and Gender 505 -- 17.4 Detection of Anger 510 -- 17.5 Applications in IVR Systems 517 -- 17.6 Discussion and Conclusion 523 -- Bibliography 525 -- Index 529 -- Permissions List 541.
Record Nr. UNINA-9910144580303321
Chichester, West Sussex, England ; , : John Wiley & Sons, , c2008
Materiale a stampa
Lo trovi qui: Univ. Federico II
Opac: Controlla la disponibilità qui
Advances in digital speech transmission / / edited by Rainer Martin, Ulrich Heute, Christiane Antweiler
Advances in digital speech transmission / / edited by Rainer Martin, Ulrich Heute, Christiane Antweiler
Edizione [1st ed.]
Pubbl/distr/stampa Chichester, West Sussex, England ; ; Hoboken, NJ, : John Wiley & Sons, c2008
Descrizione fisica 1 online resource (573 p.)
Disciplina 621.39/9
Altri autori (Persone) MartinRainer
HeuteUlrich
AntweilerChristiane
Soggetto topico Speech processing systems
Signal processing - Digital techniques
ISBN 1-281-32198-2
9786611321987
0-470-72718-7
0-470-72717-9
Formato Materiale a stampa
Livello bibliografico Monografia
Lingua di pubblicazione eng
Nota di contenuto -- List of Contributors xxi -- Preface xxvii -- 1 Introduction 1 /Rainer Martin, Ulrich Heute, Christiane Antweiler -- I Speech Quality Assessment 7 -- 2 Speech-Transmission Quality: Aspects and Assessment for Wideband vs. Narrowband Signals 9 /Ulrich Heute -- 2.1 Introduction 9 -- 2.2 Speech Signals . 10 -- 2.3 Telephone-Band Speech Signals 11 -- 2.4 Wideband Speech Signals 14 -- 2.5 Speech-Quality Assessment 25 -- 2.6 Wideband Speech-Quality Assessment 30 -- 2.7 Concluding Remarks 43 -- Bibliography 44 -- 3 Parametric Quality Assessment of Narrowband Speech in Mobile Communication Systems 51 /Marc Werner -- 3.1 Introduction 51 -- 3.2 Simulations of GSM and UMTS Speech Transmissions 58 -- 3.3 Speech Quality Measures based on Transmission Parameters 65 -- 3.4 Discussion and Conclusions 73 -- Bibliography 73 -- II Adaptive Algorithms in Acoustic Signal Processing 77 -- 4 Kalman Filtering in Acoustic Echo Control: A Smooth Ride on a Rocky Road 79 /Gerald Enzner -- 4.1 Introduction 79 -- 4.2 A Comprehensive Theory of Acoustic Echo Control 85 -- 4.3 The Kalman Filter for Conditional Mean and Covariance Estimation 90 -- 4.4 AEC Performance of the Frequency-Domain Adaptive Kalman Filter 100 -- 4.5 Discussion and Conclusions 102 -- Bibliography 103 -- 5 Noise Reduction - Statistical Analysis and Control of Musical Noise 107 /Colin Breithaupt, Rainer Martin -- 5.1 Introduction 107 -- 5.2 Speech Enhancement in the DFT Domain 109 -- 5.3 Measurement and Assessment of Unnatural Fluctuations 115 -- 5.4 Avoidance of Processing Artifacts 120 -- 5.5 Control of Spectral Fluctuations in the Cepstral Domain 123 -- 5.6 Discussion and Conclusions 128 -- 5.7 Acknowledgements 129 -- 5.8 Appendix 129 -- Bibliography 131 -- 6 Acoustic Source Localization with Microphone Arrays 135 /Nilesh Madhu, Rainer Martin -- 6.1 Introduction 135 -- 6.2 SignalModel 136 -- 6.3 Localization Approach Taxonomy 141 -- 6.4 Indirect Localization Approaches 141 -- 6.5 Direct Localization Approaches 148 -- 6.6 Evaluation of Localization Algorithms 156.
6.7 Conclusions 166 -- Bibliography 166 / /7 Multi-Channel System Identification with Perfect Sequences / Theory and Applications / 171 /Christiane Antweiler -- 7.1 Introduction 171 -- 7.2 System Identification with Perfect Sequences 174 -- 7.3 Multi-Channel System Identification 185 -- 7.4 Applications 191 -- 7.5 Discussion and Conclusions 195 -- Bibliography 195 -- III Speech Coding for Heterogeneous Networks 199 -- 8 Embedded Speech Coding: From G.711 to G.729.1 201 /Bernd Geiser, Stéphane Ragot, Hervé Taddei -- 8.1 Introduction 201 -- 8.2 Theory and Tools of Embedded Speech Coding 203 -- 8.3 Embedded Speech Coding Methods 212 -- 8.4 Standardized Embedded Speech Coders 219 -- 8.5 Network Aspects of Embedded Speech Coding 232 -- 8.6 Conclusions and Perspectives 237 -- Bibliography 238 -- 9 Backwards Compatible Wideband Telephony 249 /Peter Jax -- 9.1 Introduction 249 -- 9.2 From Narrowband Telephony to Wideband Telephony 250 -- 9.3 Stand-Alone Bandwidth Extension 254 -- 9.4 Embedded Wideband Coding Using Bandwidth Extension Techniques 257 -- 9.5 Combination of Bandwidth Extension with Watermarking 262 -- 9.6 Advanced Transmission of Highband Parameters 267 -- 9.7 Conclusions 274 -- Bibliography 274 -- IV Joint Source-Channel Coding 279 -- 10 Parameter Models and Estimators in Soft Decision Source Decoding 281 /Tim Fingscheidt -- 10.1 Introduction 281 -- 10.2 Overview to Soft Decision Source Decoding 283 -- 10.3 The Markovian Parameter Model 287 -- 10.4 Basic Extrapolative Estimators 290 -- 10.5 Joint Extrapolative Estimation of Two Different Parameters 298 -- 10.6 Extrapolative Estimation with Repeated Parameter Transmission 301 -- 10.7 Interpolative Estimation of a Parameter 304 -- 10.8 Discussion and Conclusions 307 -- Bibliography 307 -- 11 Optimal MMSE Estimation for Vector Sources with Spatially and Temporally Correlated Elements 311 /Stefan Heinen, Marc Adrat -- 11.1 Introduction 311 -- 11.2 Source Model 312 -- 11.3 Transmission Channel 316 -- 11.4 Optimal MMSE Parameter Estimator 316.
11.5 Near-Optimal MMSE Parameter Estimator 320 -- 11.6 Illustrative Comparison 323 -- 11.7 Simulation Results 325 -- 11.8 Conclusions 327 -- Bibliography 327 -- 12 Source Optimized Channel Codes & Source Controlled Channel Decoding 329 /Stefan Heinen, Thomas Hindelang -- 12.1 Introduction 329 -- 12.2 The Transmission System Used as Reference 330 -- 12.3 Source Optimized Channel Coding (SOCC) 332 -- 12.4 Source Controlled Channel Decoding (SCCD) 341 -- 12.5 Comparison of SOCC versus SCCD 357 -- 12.6 Conclusions 362 -- Bibliography 363 -- 13 Iterative Source-Channel Decoding & Turbo DeCodulation 365 /Marc Adrat, Thorsten Clevorn, Laurent Schmalen -- 13.1 Introduction 365 -- 13.2 The Key of the Turbo Principle: Extrinsic Information 366 -- 13.3 Iterative Source-Channel Decoding (ISCD) 379 -- 13.4 Turbo DeCodulation (TDeC) 387 -- 13.5 Conclusions 394 -- Bibliography 395 -- V Speech Processing in Hearing Instruments 399 -- 14 Binaural Signal Processing in Hearing Aids 401 /Volkmar Hamacher, Ulrich Kornagel, Thomas Lotter, Henning Puder -- 14.1 Introduction 401 -- 14.2 Wireless System for Hearing Aids 405 -- 14.3 Binaural Classification Systems 410 -- 14.4 Binaural Beamformer 415 -- 14.5 Blind Source Separation (BSS): An Application for a Binaural Directional Microphone Array in Hearing Aids 422 -- 14.6 Conclusions 427 -- Bibliography 428 -- 15 Auditory-profile-based Physical Evaluation of Multi-microphone Noise Reduction Techniques in Hearing Instruments 431 /Koen Eneman, Arne Leijon, Simon Doclo, Ann Spriet, Marc Moonen, Jan Wouters -- 15.1 Introduction 431 -- 15.2 Multi-microphone Noise Reduction in Hearing Instruments 434 -- 15.3 Auditory-profile-based Physical Evaluation 441 -- 15.4 Test Conditions 449 -- 15.5 Simulation Results 450 -- 15.6 Discussion 452 -- 15.7 Conclusions 455 -- Bibliography 456 -- VI Speech Processing for Human / Machine Interfaces 459 -- 16 Automatic Speech Recognition in Adverse Acoustic Conditions 461 /Hans-GŠunter Hirsch -- 16.1 Introduction 461.
16.2 Structure of Speech Recognition Systems 462 -- 16.3 Acoustic Scenarios during Speech Input 468 -- 16.4 Improving the Recognition Performance in Adverse Conditions 476 -- 16.5 Conclusions 493 -- Bibliography 494 -- 17 Speaker Classification for Next-Generation Voice-Dialog Systems 497 /Felix Burkhardt, Florian Metze, Joachim Stegmann -- 17.1 Introduction 497 -- 17.2 Speaker Classification 498 -- 17.3 Detection of Age and Gender 505 -- 17.4 Detection of Anger 510 -- 17.5 Applications in IVR Systems 517 -- 17.6 Discussion and Conclusion 523 -- Bibliography 525 -- Index 529 -- Permissions List 541.
Record Nr. UNINA-9910811945903321
Chichester, West Sussex, England ; ; Hoboken, NJ, : John Wiley & Sons, c2008
Materiale a stampa
Lo trovi qui: Univ. Federico II
Opac: Controlla la disponibilità qui
Digital speech transmission [[electronic resource] ] : enhancement, coding and error concealment / / Peter Vary, Rainer Martin
Digital speech transmission [[electronic resource] ] : enhancement, coding and error concealment / / Peter Vary, Rainer Martin
Autore Vary Peter
Pubbl/distr/stampa Chichester, England ; ; Hoboken, NJ, : John Wiley, c2006
Descrizione fisica 1 online resource (645 p.)
Disciplina 621.399
Altri autori (Persone) MartinRainer
Soggetto topico Speech processing systems
Signal processing - Digital techniques
Error-correcting codes (Information theory)
ISBN 1-280-60611-8
9786610606115
0-470-03174-3
0-470-03175-1
Formato Materiale a stampa
Livello bibliografico Monografia
Lingua di pubblicazione eng
Nota di contenuto Digital Speech Transmission; Contents; Preface; 1 Introduction; 2 Models of Speech Production and Hearing; 2.1 Organs of Speech Production; 2.2 Characteristics of Speech Signals; 2.3 Model of Speech Production; 2.3.1 Acoustic Tube Model of the Vocal Tract; 2.3.2 Digital All-Pole Model of the Vocal Tract; 2.4 Anatomy of Hearing; 2.5 Psychoacoustic Properties of the Auditory Organ; 2.5.1 Hearing and Loudness; 2.5.2 Spectral Resolution; 2.5.3 Masking; Bibliography; 3 Spectral Transformations; 3.1 Fourier Transform of Continuous Signals; 3.2 Fourier Transform of Discrete Signals
3.3 Linear Shift Invariant Systems3.3.1 Frequency Response of LSI Systems; 3.4 The z-transform; 3.4.1 Relation to FT; 3.4.2 Properties of the ROC; 3.4.3 Inverse z-transform; 3.4.4 z-transform Analysis of LSI Systems; 3.5 The Discrete Fourier Transform; 3.5.1 Linear and Cyclic Convolution; 3.5.2 The DFT of Windowed Sequences; 3.5.3 Spectral Resolution and Zero Padding; 3.5.4 Fast Computation of the DFT: The FFT; 3.5.5 Radix-2 Decimation-in-Time FFT; 3.6 Fast Convolution; 3.6.1 Fast Convolution of Long Sequences; 3.6.2 Fast Convolution by Overlap-Add; 3.6.3 Fast Convolution by Overlap-Save
3.7 Cepstral Analysis3.7.1 Complex Cepstrum; 3.7.2 Real Cepstrum; 3.7.3 Applications of the Cepstrum; Bibliography; 4 Filter Banks for Spectral Analysis and Synthesis; 4.1 Spectral Analysis Using Narrowband Filters; 4.1.1 Short-Term Spectral Analyzer; 4.1.2 Prototype Filter Design for the Analysis Filter Bank; 4.1.3 Short-Term Spectral Synthesizer; 4.1.4 Short-Term Spectral Analysis and Synthesis; 4.1.5 Prototype Filter Design for the Analysis-Synthesis Filter Bank; 4.1.6 Filter Bank Interpretation of the DFT; 4.2 Polyphase Network Filter Banks; 4.2.1 PPN Analysis Filter Bank
4.2.2 PPN Synthesis Filter Bank4.3 Quadrature Mirror Filter Banks; 4.3.1 Analysis-Synthesis Filter Bank; 4.3.2 Compensation of Aliasing and Signal Reconstruction; 4.3.3 Efficient Implementation; Bibliography; 5 Stochastic Signals and Estimation; 5.1 Basic Concepts; 5.1.1 Random Events and Probability; 5.1.2 Conditional Probabilities; 5.1.3 Random Variables; 5.1.4 Probability Distributions and Probability Density Functions; 5.1.5 Conditional PDFs; 5.2 Expectations and Moments; 5.2.1 Conditional Expectations and Moments; 5.2.2 Examples; 5.2.3 Transformation of a Random Variable
5.2.4 Relative Frequencies and Histograms5.3 Bivariate Statistics; 5.3.1 Marginal Densities; 5.3.2 Expectations and Moments; 5.3.3 Uncorrelatedness and Statistical Independence; 5.3.4 Examples of Bivariate PDFs; 5.3.5 Functions of Two Random Variables; 5.4 Probability and Information; 5.4.1 Entropy; 5.4.2 Kullback-Leibler Divergence; 5.4.3 Mutual Information; 5.5 Multivariate Statistics; 5.5.1 MultivariateGaussian Distribution; 5.5.2 χ2-distribution; 5.6 Stochastic Processes; 5.6.1 Stationary Processes; 5.6.2 Auto-correlation and Auto-covariance Functions
5.6.3 Cross-correlation and Cross-covariance Functions
Record Nr. UNINA-9910143561303321
Vary Peter  
Chichester, England ; ; Hoboken, NJ, : John Wiley, c2006
Materiale a stampa
Lo trovi qui: Univ. Federico II
Opac: Controlla la disponibilità qui
Digital speech transmission [[electronic resource] ] : enhancement, coding and error concealment / / Peter Vary, Rainer Martin
Digital speech transmission [[electronic resource] ] : enhancement, coding and error concealment / / Peter Vary, Rainer Martin
Autore Vary Peter
Pubbl/distr/stampa Chichester, England ; ; Hoboken, NJ, : John Wiley, c2006
Descrizione fisica 1 online resource (645 p.)
Disciplina 621.399
Altri autori (Persone) MartinRainer
Soggetto topico Speech processing systems
Signal processing - Digital techniques
Error-correcting codes (Information theory)
ISBN 1-280-60611-8
9786610606115
0-470-03174-3
0-470-03175-1
Formato Materiale a stampa
Livello bibliografico Monografia
Lingua di pubblicazione eng
Nota di contenuto Digital Speech Transmission; Contents; Preface; 1 Introduction; 2 Models of Speech Production and Hearing; 2.1 Organs of Speech Production; 2.2 Characteristics of Speech Signals; 2.3 Model of Speech Production; 2.3.1 Acoustic Tube Model of the Vocal Tract; 2.3.2 Digital All-Pole Model of the Vocal Tract; 2.4 Anatomy of Hearing; 2.5 Psychoacoustic Properties of the Auditory Organ; 2.5.1 Hearing and Loudness; 2.5.2 Spectral Resolution; 2.5.3 Masking; Bibliography; 3 Spectral Transformations; 3.1 Fourier Transform of Continuous Signals; 3.2 Fourier Transform of Discrete Signals
3.3 Linear Shift Invariant Systems3.3.1 Frequency Response of LSI Systems; 3.4 The z-transform; 3.4.1 Relation to FT; 3.4.2 Properties of the ROC; 3.4.3 Inverse z-transform; 3.4.4 z-transform Analysis of LSI Systems; 3.5 The Discrete Fourier Transform; 3.5.1 Linear and Cyclic Convolution; 3.5.2 The DFT of Windowed Sequences; 3.5.3 Spectral Resolution and Zero Padding; 3.5.4 Fast Computation of the DFT: The FFT; 3.5.5 Radix-2 Decimation-in-Time FFT; 3.6 Fast Convolution; 3.6.1 Fast Convolution of Long Sequences; 3.6.2 Fast Convolution by Overlap-Add; 3.6.3 Fast Convolution by Overlap-Save
3.7 Cepstral Analysis3.7.1 Complex Cepstrum; 3.7.2 Real Cepstrum; 3.7.3 Applications of the Cepstrum; Bibliography; 4 Filter Banks for Spectral Analysis and Synthesis; 4.1 Spectral Analysis Using Narrowband Filters; 4.1.1 Short-Term Spectral Analyzer; 4.1.2 Prototype Filter Design for the Analysis Filter Bank; 4.1.3 Short-Term Spectral Synthesizer; 4.1.4 Short-Term Spectral Analysis and Synthesis; 4.1.5 Prototype Filter Design for the Analysis-Synthesis Filter Bank; 4.1.6 Filter Bank Interpretation of the DFT; 4.2 Polyphase Network Filter Banks; 4.2.1 PPN Analysis Filter Bank
4.2.2 PPN Synthesis Filter Bank4.3 Quadrature Mirror Filter Banks; 4.3.1 Analysis-Synthesis Filter Bank; 4.3.2 Compensation of Aliasing and Signal Reconstruction; 4.3.3 Efficient Implementation; Bibliography; 5 Stochastic Signals and Estimation; 5.1 Basic Concepts; 5.1.1 Random Events and Probability; 5.1.2 Conditional Probabilities; 5.1.3 Random Variables; 5.1.4 Probability Distributions and Probability Density Functions; 5.1.5 Conditional PDFs; 5.2 Expectations and Moments; 5.2.1 Conditional Expectations and Moments; 5.2.2 Examples; 5.2.3 Transformation of a Random Variable
5.2.4 Relative Frequencies and Histograms5.3 Bivariate Statistics; 5.3.1 Marginal Densities; 5.3.2 Expectations and Moments; 5.3.3 Uncorrelatedness and Statistical Independence; 5.3.4 Examples of Bivariate PDFs; 5.3.5 Functions of Two Random Variables; 5.4 Probability and Information; 5.4.1 Entropy; 5.4.2 Kullback-Leibler Divergence; 5.4.3 Mutual Information; 5.5 Multivariate Statistics; 5.5.1 MultivariateGaussian Distribution; 5.5.2 χ2-distribution; 5.6 Stochastic Processes; 5.6.1 Stationary Processes; 5.6.2 Auto-correlation and Auto-covariance Functions
5.6.3 Cross-correlation and Cross-covariance Functions
Record Nr. UNINA-9910830553603321
Vary Peter  
Chichester, England ; ; Hoboken, NJ, : John Wiley, c2006
Materiale a stampa
Lo trovi qui: Univ. Federico II
Opac: Controlla la disponibilità qui
Digital speech transmission : enhancement, coding and error concealment / / Peter Vary, Rainer Martin
Digital speech transmission : enhancement, coding and error concealment / / Peter Vary, Rainer Martin
Autore Vary Peter
Pubbl/distr/stampa Chichester, England ; ; Hoboken, NJ, : John Wiley, c2006
Descrizione fisica 1 online resource (645 p.)
Disciplina 621.39/9
Altri autori (Persone) MartinRainer
Soggetto topico Speech processing systems
Signal processing - Digital techniques
Error-correcting codes (Information theory)
ISBN 1-280-60611-8
9786610606115
0-470-03174-3
0-470-03175-1
Formato Materiale a stampa
Livello bibliografico Monografia
Lingua di pubblicazione eng
Nota di contenuto Digital Speech Transmission; Contents; Preface; 1 Introduction; 2 Models of Speech Production and Hearing; 2.1 Organs of Speech Production; 2.2 Characteristics of Speech Signals; 2.3 Model of Speech Production; 2.3.1 Acoustic Tube Model of the Vocal Tract; 2.3.2 Digital All-Pole Model of the Vocal Tract; 2.4 Anatomy of Hearing; 2.5 Psychoacoustic Properties of the Auditory Organ; 2.5.1 Hearing and Loudness; 2.5.2 Spectral Resolution; 2.5.3 Masking; Bibliography; 3 Spectral Transformations; 3.1 Fourier Transform of Continuous Signals; 3.2 Fourier Transform of Discrete Signals
3.3 Linear Shift Invariant Systems3.3.1 Frequency Response of LSI Systems; 3.4 The z-transform; 3.4.1 Relation to FT; 3.4.2 Properties of the ROC; 3.4.3 Inverse z-transform; 3.4.4 z-transform Analysis of LSI Systems; 3.5 The Discrete Fourier Transform; 3.5.1 Linear and Cyclic Convolution; 3.5.2 The DFT of Windowed Sequences; 3.5.3 Spectral Resolution and Zero Padding; 3.5.4 Fast Computation of the DFT: The FFT; 3.5.5 Radix-2 Decimation-in-Time FFT; 3.6 Fast Convolution; 3.6.1 Fast Convolution of Long Sequences; 3.6.2 Fast Convolution by Overlap-Add; 3.6.3 Fast Convolution by Overlap-Save
3.7 Cepstral Analysis3.7.1 Complex Cepstrum; 3.7.2 Real Cepstrum; 3.7.3 Applications of the Cepstrum; Bibliography; 4 Filter Banks for Spectral Analysis and Synthesis; 4.1 Spectral Analysis Using Narrowband Filters; 4.1.1 Short-Term Spectral Analyzer; 4.1.2 Prototype Filter Design for the Analysis Filter Bank; 4.1.3 Short-Term Spectral Synthesizer; 4.1.4 Short-Term Spectral Analysis and Synthesis; 4.1.5 Prototype Filter Design for the Analysis-Synthesis Filter Bank; 4.1.6 Filter Bank Interpretation of the DFT; 4.2 Polyphase Network Filter Banks; 4.2.1 PPN Analysis Filter Bank
4.2.2 PPN Synthesis Filter Bank4.3 Quadrature Mirror Filter Banks; 4.3.1 Analysis-Synthesis Filter Bank; 4.3.2 Compensation of Aliasing and Signal Reconstruction; 4.3.3 Efficient Implementation; Bibliography; 5 Stochastic Signals and Estimation; 5.1 Basic Concepts; 5.1.1 Random Events and Probability; 5.1.2 Conditional Probabilities; 5.1.3 Random Variables; 5.1.4 Probability Distributions and Probability Density Functions; 5.1.5 Conditional PDFs; 5.2 Expectations and Moments; 5.2.1 Conditional Expectations and Moments; 5.2.2 Examples; 5.2.3 Transformation of a Random Variable
5.2.4 Relative Frequencies and Histograms5.3 Bivariate Statistics; 5.3.1 Marginal Densities; 5.3.2 Expectations and Moments; 5.3.3 Uncorrelatedness and Statistical Independence; 5.3.4 Examples of Bivariate PDFs; 5.3.5 Functions of Two Random Variables; 5.4 Probability and Information; 5.4.1 Entropy; 5.4.2 Kullback-Leibler Divergence; 5.4.3 Mutual Information; 5.5 Multivariate Statistics; 5.5.1 MultivariateGaussian Distribution; 5.5.2 χ2-distribution; 5.6 Stochastic Processes; 5.6.1 Stationary Processes; 5.6.2 Auto-correlation and Auto-covariance Functions
5.6.3 Cross-correlation and Cross-covariance Functions
Record Nr. UNINA-9910877487603321
Vary Peter  
Chichester, England ; ; Hoboken, NJ, : John Wiley, c2006
Materiale a stampa
Lo trovi qui: Univ. Federico II
Opac: Controlla la disponibilità qui
Digital Speech Transmission and Enhancement
Digital Speech Transmission and Enhancement
Autore Vary Peter
Edizione [2nd ed.]
Pubbl/distr/stampa Newark : , : John Wiley & Sons, Incorporated, , 2023
Descrizione fisica 1 online resource (595 pages)
Disciplina 006.454
Altri autori (Persone) MartinRainer
Collana IEEE Press Series
ISBN 1-119-06099-0
1-119-06097-4
Formato Materiale a stampa
Livello bibliografico Monografia
Lingua di pubblicazione eng
Nota di contenuto Cover -- Title Page -- Copyright -- Contents -- Preface -- Chapter 1 Introduction -- Chapter 2 Models of Speech Production and Hearing -- 2.1 Sound Waves -- 2.2 Organs of Speech Production -- 2.3 Characteristics of Speech Signals -- 2.4 Model of Speech Production -- 2.4.1 Acoustic Tube Model of the Vocal Tract -- 2.4.2 Discrete Time All‐Pole Model of the Vocal Tract -- 2.5 Anatomy of Hearing -- 2.6 Psychoacoustic Properties of the Auditory System -- 2.6.1 Hearing and Loudness -- 2.6.2 Spectral Resolution -- 2.6.3 Masking -- 2.6.4 Spatial Hearing -- 2.6.4.1 Head‐Related Impulse Responses and Transfer Functions -- 2.6.4.2 Law of The First Wavefront -- References -- Chapter 3 Spectral Transformations -- 3.1 Fourier Transform of Continuous Signals -- 3.2 Fourier Transform of Discrete Signals -- 3.3 Linear Shift Invariant Systems -- 3.3.1 Frequency Response of LSI Systems -- 3.4 The z‐transform -- 3.4.1 Relation to Fourier Transform -- 3.4.2 Properties of the ROC -- 3.4.3 Inverse z‐Transform -- 3.4.4 z‐Transform Analysis of LSI Systems -- 3.5 The Discrete Fourier Transform -- 3.5.1 Linear and Cyclic Convolution -- 3.5.2 The DFT of Windowed Sequences -- 3.5.3 Spectral Resolution and Zero Padding -- 3.5.4 The Spectrogram -- 3.5.5 Fast Computation of the DFT: The FFT -- 3.5.6 Radix‐2 Decimation‐in‐Time FFT -- 3.6 Fast Convolution -- 3.6.1 Fast Convolution of Long Sequences -- 3.6.2 Fast Convolution by Overlap‐Add -- 3.6.3 Fast Convolution by Overlap‐Save -- 3.7 Analysis-Modification-Synthesis Systems -- 3.8 Cepstral Analysis -- 3.8.1 Complex Cepstrum -- 3.8.2 Real Cepstrum -- 3.8.3 Applications of the Cepstrum -- 3.8.3.1 Construction of Minimum‐Phase Sequences -- 3.8.3.2 Deconvolution by Cepstral Mean Subtraction -- 3.8.3.3 Computation of the Spectral Distortion Measure -- 3.8.3.4 Fundamental Frequency Estimation -- References.
Chapter 4 Filter Banks for Spectral Analysis and Synthesis -- 4.1 Spectral Analysis Using Narrowband Filters -- 4.1.1 Short‐Term Spectral Analyzer -- 4.1.2 Prototype Filter Design for the Analysis Filter Bank -- 4.1.3 Short‐Term Spectral Synthesizer -- 4.1.4 Short‐Term Spectral Analysis and Synthesis -- 4.1.5 Prototype Filter Design for the Analysis-Synthesis filter bank -- 4.1.6 Filter Bank Interpretation of the DFT -- 4.2 Polyphase Network Filter Banks -- 4.2.1 PPN Analysis Filter Bank -- 4.2.2 PPN Synthesis Filter Bank -- 4.3 Quadrature Mirror Filter Banks -- 4.3.1 Analysis-Synthesis Filter Bank -- 4.3.2 Compensation of Aliasing and Signal Reconstruction -- 4.3.3 Efficient Implementation -- 4.4 Filter Bank Equalizer -- 4.4.1 The Reference Filter Bank -- 4.4.2 Uniform Frequency Resolution -- 4.4.3 Adaptive Filter Bank Equalizer: Gain Computation -- 4.4.3.1 Conventional Spectral Subtraction -- 4.4.3.2 Filter Bank Equalizer -- 4.4.4 Non‐uniform Frequency Resolution -- 4.4.5 Design Aspects & -- Implementation -- References -- Chapter 5 Stochastic Signals and Estimation -- 5.1 Basic Concepts -- 5.1.1 Random Events and Probability -- 5.1.2 Conditional Probabilities -- 5.1.3 Random Variables -- 5.1.4 Probability Distributions and Probability Density Functions -- 5.1.5 Conditional PDFs -- 5.2 Expectations and Moments -- 5.2.1 Conditional Expectations and Moments -- 5.2.2 Examples -- 5.2.2.1 The Uniform Distribution -- 5.2.2.2 The Gaussian Density -- 5.2.2.3 The Exponential Density -- 5.2.2.4 The Laplace Density -- 5.2.2.5 The Gamma Density -- 5.2.2.6 χ2‐Distribution -- 5.2.3 Transformation of a Random Variable -- 5.2.4 Relative Frequencies and Histograms -- 5.3 Bivariate Statistics -- 5.3.1 Marginal Densities -- 5.3.2 Expectations and Moments -- 5.3.3 Uncorrelatedness and Statistical Independence -- 5.3.4 Examples of Bivariate PDFs.
5.3.4.1 The Bivariate Uniform Density -- 5.3.4.2 The Bivariate Gaussian Density -- 5.3.5 Functions of Two Random Variables -- 5.4 Probability and Information -- 5.4.1 Entropy -- 5.4.2 Kullback-Leibler Divergence -- 5.4.3 Cross‐Entropy -- 5.4.4 Mutual Information -- 5.5 Multivariate Statistics -- 5.5.1 Multivariate Gaussian Distribution -- 5.5.2 Gaussian Mixture Models -- 5.6 Stochastic Processes -- 5.6.1 Stationary Processes -- 5.6.2 Auto‐Correlation and Auto‐Covariance Functions -- 5.6.3 Cross‐Correlation and Cross‐Covariance Functions -- 5.6.4 Markov Processes -- 5.6.5 Multivariate Stochastic Processes -- 5.7 Estimation of Statistical Quantities by Time Averages -- 5.7.1 Ergodic Processes -- 5.7.2 Short‐Time Stationary Processes -- 5.8 Power Spectrum and its Estimation -- 5.8.1 White Noise -- 5.8.2 The Periodogram -- 5.8.3 Smoothed Periodograms -- 5.8.3.1 Non Recursive Smoothing in Time -- 5.8.3.2 Recursive Smoothing in Time -- 5.8.3.3 Log‐Mel Filter Bank Features -- 5.8.4 Power Spectra and Linear Shift‐Invariant Systems -- 5.9 Statistical Properties of Speech Signals -- 5.10 Statistical Properties of DFT Coefficients -- 5.10.1 Asymptotic Statistical Properties -- 5.10.2 Signal‐Plus‐Noise Model -- 5.10.3 Statistics of DFT Coefficients for Finite Frame Lengths -- 5.11 Optimal Estimation -- 5.11.1 MMSE Estimation -- 5.11.2 Estimation of Discrete Random Variables -- 5.11.3 Optimal Linear Estimator -- 5.11.4 The Gaussian Case -- 5.11.5 Joint Detection and Estimation -- 5.12 Non‐Linear Estimation with Deep Neural Networks -- 5.12.1 Basic Network Components -- 5.12.1.1 The Perceptron -- 5.12.1.2 Convolutional Neural Network -- 5.12.2 Basic DNN Structures -- 5.12.2.1 Fully‐Connected Feed‐Forward Network -- 5.12.2.2 Autoencoder Networks -- 5.12.2.3 Recurrent Neural Networks -- 5.12.2.4 Time Delay, Wavenet, and Transformer Networks.
5.12.2.5 Training of Neural Networks -- 5.12.2.6 Stochastic Gradient Descent (SGD) -- 5.12.2.7 Adaptive Moment Estimation Method (ADAM) -- References -- Chapter 6 Linear Prediction -- 6.1 Vocal Tract Models and Short‐Term Prediction -- 6.1.1 All‐Zero Model -- 6.1.2 All‐Pole Model -- 6.1.3 Pole‐Zero Model -- 6.2 Optimal Prediction Coefficients for Stationary Signals -- 6.2.1 Optimum Prediction -- 6.2.2 Spectral Flatness Measure -- 6.3 Predictor Adaptation -- 6.3.1 Block‐Oriented Adaptation -- 6.3.1.1 Auto‐Correlation Method -- 6.3.1.2 Covariance Method -- 6.3.1.3 Levinson-Durbin Algorithm -- 6.3.2 Sequential Adaptation -- 6.4 Long‐Term Prediction -- References -- Chapter 7 Quantization -- 7.1 Analog Samples and Digital Representation -- 7.2 Uniform Quantization -- 7.3 Non‐uniform Quantization -- 7.4 Optimal Quantization -- 7.5 Adaptive Quantization -- 7.6 Vector Quantization -- 7.6.1 Principle -- 7.6.2 The Complexity Problem -- 7.6.3 Lattice Quantization -- 7.6.4 Design of Optimal Vector Code Books -- 7.6.5 Gain-Shape Vector Quantization -- 7.7 Quantization of the Predictor Coefficients -- 7.7.1 Scalar Quantization of the LPC Coefficients -- 7.7.2 Scalar Quantization of the Reflection Coefficients -- 7.7.3 Scalar Quantization of the LSF Coefficients -- References -- Chapter 8 Speech Coding -- 8.1 Speech‐Coding Categories -- 8.2 Model‐Based Predictive Coding -- 8.3 Linear Predictive Waveform Coding -- 8.3.1 First‐Order DPCM -- 8.3.2 Open‐Loop and Closed‐Loop Prediction -- 8.3.3 Quantization of the Residual Signal -- 8.3.3.1 Quantization with Open‐Loop Prediction -- 8.3.3.2 Quantization with Closed‐Loop Prediction -- 8.3.3.3 Spectral Shaping of the Quantization Error -- 8.3.4 ADPCM with Sequential Adaptation -- 8.4 Parametric Coding -- 8.4.1 Vocoder Structures -- 8.4.2 LPC Vocoder -- 8.5 Hybrid Coding -- 8.5.1 Basic Codec Concepts.
8.5.1.1 Scalar Quantization of the Residual Signal -- 8.5.1.2 Vector Quantization of the Residual Signal -- 8.5.2 Residual Signal Coding: RELP -- 8.5.3 Analysis by Synthesis: CELP -- 8.5.3.1 Principle -- 8.5.3.2 Fixed Code Book -- 8.5.3.3 Long‐Term Prediction, Adaptive Code Book -- 8.6 Adaptive Postfiltering -- 8.7 Speech Codec Standards: Selected Examples -- 8.7.1 GSM Full‐Rate Codec -- 8.7.2 EFR Codec -- 8.7.3 Adaptive Multi‐Rate Narrowband Codec (AMR‐NB) -- 8.7.4 ITU‐T/G.722: 7 kHz Audio Coding within 64 kbit/s -- 8.7.5 Adaptive Multi‐Rate Wideband Codec (AMR‐WB) -- 8.7.6 Codec for Enhanced Voice Services (EVS) -- 8.7.7 Opus Codec IETF RFC 6716 -- References -- Chapter 9 Concealment of Erroneous or Lost Frames -- 9.1 Concepts for Error Concealment -- 9.1.1 Error Concealment by Hard Decision Decoding -- 9.1.2 Error Concealment by Soft Decision Decoding -- 9.1.3 Parameter Estimation -- 9.1.3.1 MAP Estimation -- 9.1.3.2 MS Estimation -- 9.1.4 The A Posteriori Probabilities -- 9.1.4.1 The A Priori Knowledge -- 9.1.4.2 The Parameter Distortion Probabilities -- 9.1.5 Example: Hard Decision vs. Soft Decision -- 9.2 Examples of Error Concealment Standards -- 9.2.1 Substitution and Muting of Lost Frames -- 9.2.2 AMR Codec: Substitution and Muting of Lost Frames -- 9.2.3 EVS Codec: Concealment of Lost Packets -- 9.3 Further Improvements -- References -- Chapter 10 Bandwidth Extension of Speech Signals -- 10.1 BWE Concepts -- 10.2 BWE using the Model of Speech Production -- 10.2.1 Extension of the Excitation Signal -- 10.2.2 Spectral Envelope Estimation -- 10.2.2.1 Minimum Mean Square Error Estimation -- 10.2.2.2 Conditional Maximum A Posteriori Estimation -- 10.2.2.3 Extensions -- 10.2.2.4 Simplifications -- 10.2.3 Energy Envelope Estimation -- 10.3 Speech Codecs with Integrated BWE -- 10.3.1 BWE in the GSM Full‐Rate Codec.
10.3.2 BWE in the AMR Wideband Codec.
Record Nr. UNINA-9910829806303321
Vary Peter  
Newark : , : John Wiley & Sons, Incorporated, , 2023
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